Method and system for processing voice traffic from a multi-channel link into a VoIP network over a broadband network

ABSTRACT

A microprocessor and a DSP are coupled with a cable modem and an IAD framer to terminate a DS1 link at a media terminal adaptor device. The processor performs call control operations and the DSP performs signal processing, such that call processing occurs at the device. The framer extracts or inserts call control information from robbed-bit signaling bits. Data streams from the DS1 link are formatted/packetized according to a protocol used by a VoIP network. VoIP is used to transmit the packetized call from the cable modem portion to a CMTS over a broadband network. QoS attributes are implemented on the derived data streams as would be done for calls generated by a standard PacketCable EMTA.

CROSS REFERENCE TO RELATED APPLICATION

This application claims priority under 35 U.S.C. 119(e) to the filingdate of Ansley, U.S. provisional patent application No. 60/498,787entitled “DS1 Media Terminal Adaptor”, which was filed Aug. 29, 2003,and is incorporated herein by reference in its entirety.

FIELD OF THE INVENTION

The present invention relates generally to broadband communication, andmore particularly to a method for processing and providing voice trafficfrom a multi-channel link over an IP network.

BACKGROUND

Community antenna television (“CATV”) networks have been used for morethen four decades to deliver television programming to a large number ofsubscribers. Increasingly, CATV networks are used by providers toprovide data services to subscribers. For example, cable modems used ina broadband cable modem termination system (“CMTS”) are capable oftransmitting and receiving Internet data using the Data Over CableService Interface Specification (“DOCSIS”) protocol. DOCSIS provides astandard that allows network devices made by different vendors tocommunication with one another.

Similar to DOCSIS, which is administered by Cable TelevisionLaboratories, Inc. (CableLabs®), “PacketCable™ is a CableLabs-ledinitiative aimed at developing interoperable interface specificationsfor delivering advanced, real-time multimedia services over two-waycable plant. Built on top of the industry's highly successful cablemodem infrastructure, PacketCable networks will use Internet protocol(IP) technology to enable a wide range of multimedia services, such asIP telephony, multimedia conferencing, interactive gaming, and generalmultimedia applications.” See www.packetcable.com. DOCSIS andPacketCable are protocol standards known in the art and do not requirefurther discussion of the basic functioning thereof. However, it will beappreciated that, although DOCSIS and PacketCable are currentlyconsidered industry standards, other protocol standards may becomepredominant over time. Thus, for purposes of discussion herein, DOCSISmay be generically referred to as a ‘data protocol’ and PacketCable as a‘multimedia protocol.’

A multimedia protocol can be used in conjunction with the Internet toprovide services, such as, for example, a voice call that emulates aplain old telephone service (“POTS”) telephone call. Another servicecommon in the pre-existing Telecommunication marketplace is DS1 or E1multi-channel transport service. Several vendors have developed devicesthat allow multi-channel TDM services, such as DS1 service or E1service, to be transported over packet-switched networks, such as theInternet. These devices are known in the art as Integrated AccessDevices (“IAD”). As shown in FIG. 1, a public branch exchange (“PBX”) 2is often served by a DS1 link 4, DS1 technology being known in the art.The DS1 4 and PBX 2 arrangement may typically be used to supportmultiple call circuits to an office building, for example. IAD 6provides an interface between the DS1 with its TDM formatting and acable modem 8 with its packet-switched technology. Thus, the 24 channelsof DS1 4 are formatted by IAD 6 before being transported by modem 8 toCMTS 10. Link 12 between modem 8 and CMTS 10 represent a coaxial or HFCnetwork, for example.

After data contained in the formatted 24 channels has been transportedacross network 14, which is preferably the Internet or a similarpacket-switched network, IAD 16 removes the formatting performed at IAD6. The 24 channel data streams are then provided to a conventionaltelephony central office, which routes calls that originated at PBX 2 toa POTS network. Thus, conventional POTS telephony calls placed (orreceived) from/at PBX 2 are packaged and formatted at IAD 6 before beingtransported across network 14 before being unformatted/unpackaged at IAD16. This process is typically referred to in the art as ‘transport’because the continuous TDM digital data stream bearing the call trafficbetween central office 18 and PBX 2 is essentially packed into IPpackets at one end and sent to the other end, where the continuousdigital data stream is essentially ‘unpacked’ from the packetized streamand reformed into the original continuous data stream that existedbefore being converted.

While transport of the 24 DS1 channels provides a means for an operatorto provide traditional telecommunication services over a packetizedcable network, thus providing competition to the traditional telephonecompanies, implementation of the system in FIG. 1 can result ininefficient use of cable network bandwidth. For example, since transportof signals provides at the end point the data stream that was injectedat the starting point, a caller's audio silence still results inbandwidth being allocated to the call's channel. Even though there is noinformation being transmitted or received, and thus no bandwidth neededto carry useful information, the same amount of bandwidth between IAD 6and IAD 16 is reserved as is used when the DS1 channel is carrying denseaudio information. This is bandwidth that is not available forallocation to other users over cable network 12 and packet-switchednetwork 14. Furthermore, IADs 6 and 16 only package the voice callinformation into a format for transporting between modem 8 and CMTS andacross the packet-switched network 14 to the other IAD. Accordingly,call processing features are not performed by IADs, and must beperformed by legacy POTS network switches. Alternatively, the operatormust purchase additional equipment that will translate signals at a PBXDS1 into VoIP signals.

Thus, there is a need in the art for a method and system for using acable data network and the internet to support multi-channeltelecommunications services, as well as other multimedia signals, thatminimize bandwidth waste and minimize the cost to provide these servicesusing industry preferred networking and VoIP technologies.

SUMMARY

A DS1 link is terminated at a customer premise equipment media terminaladaptor (“CPEMTA”) device. The CPEMTA includes a DS1 framer andassociated processing circuitry(essentially an IAD), a cable modem and adigital signal processor. The framer extracts and inserts robbed-bitsignaling (or data links in ISDN PRI or E1 CAS signal structures, bothknown in the art) that are provided to associated processing circuitryto extract call control information. For example, a particularrobbed-bit pattern can indicate that minimal bandwidth is to be used, asthat particular channel of the multi-channel data link is idle. Thisinformation is used to drive an interface into the operator's VoIPnetwork. The data channels from the multi-channel data link are thenformatted according to the protocol used by the VoIP network, such as,for example, PacketCable, SIP, or H.323, all known in the art. Thus,from the CPEMTA on in the network, the traditional circuit-switched callhas been converted into a VoIP call. PacketCable, SIP, H.323, or othervoice over Internet protocol (“VoIP”) for example, may be used totransmit the packetized call from the cable modem portion to a CMTS overa broadband network. By integrating the multichannel conversion with aCM, quality of service QoS attributes can be implemented on the deriveddata streams, just as they would be done for calls generated by astandard PacketCable EMTA.

While the various elements of this design are now sometimes presentwithin a network, they are distributed in such a way as to prevent thisapplication from being implemented. The framer in an IAD cannot providethe level of voice processing required by VoIP; a DSP is necessary. TheDSP banks present in Gateway devices can perform the voice processingdescribed, but they are located too far back in the network for anybandwidth conservation advantages. They are also typically sized tohandle 28 or more DS1 data streams, which would not be efficient at theHFC access edge of the network.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates a system for transporting multi-channel data streamsover a broadband network.

FIG. 2 illustrates a system for providing telephony and othermulti-media services over a packetized network using a VoIP protocol.

FIG. 3 illustrates a method for providing telephony and othermulti-media services over a packetized network using a VoIP protocol.

FIG. 4 illustrates a method for initiating a call from a PBX in which aconnected multi-channel link is terminated at a media terminal adaptor.

DETAILED DESCRIPTION

As a preliminary matter, it will be readily understood by those personsskilled in the art that the present invention is susceptible of broadutility and application. Many methods, embodiments and adaptations ofthe present invention other than those herein described, as well as manyvariations, modifications, and equivalent arrangements, will be apparentfrom or reasonably suggested by the present invention and the followingdescription thereof, without departing from the substance or scope ofthe present invention.

Accordingly, while the present invention has been described herein indetail in relation to preferred embodiments, it is to be understood thatthis disclosure is only illustrative and exemplary of the presentinvention and is made merely for the purposes of providing a full andenabling disclosure of the invention. This disclosure is not intendednor is to be construed to limit the present invention or otherwise toexclude other embodiments, adaptations, variations, modifications andequivalent arrangements, the present invention being limited only by theclaims appended hereto and the equivalents thereof.

Turning now to the figures, FIG. 2 illustrates a system 22 for providingvoice and other multimedia services over a cable network 12 and IPnetwork 14. Voice calls placed or received at phones served by PBX 2 areconnected by DS1 link 4. Customer premise equipment (“CPE”) mediaterminal adaptor (“CPEMTA”) 24 processes the multi-channel time divisionmultiplexing (“TDM”) data link and processes the TDM voice data withpulse code modulation (“PCM”) encoding for use with a VoIP system. Theprocesses performed on the TDM data may include, but are not limited to,echo cancellation, codec compression and decompression, etc, theprocessed data being converted into VoIP packets that are packaged fortransmission over network 12. The reverse processes are also performedon packets received, including unpackaging packetized VoIP informationwhen received from network 12.

Call processing information is also translated between the twodomains—TDM and VoIP—by CPEMTA 24. CPEMTA 24 includes a cable modemportion 26, a processor portion 29, a Digital Signal Processor (“DSP”)portion 30, and a framer portion 28. Framer portion 28 handles thebi-directional interface with the DS1 link 4. The processor portion 29processes call control associated with the voice calls (or other multimedia signals). Call control, as known in the art, typically includesthe sending of messages to establish, maintain or release a callconnection. Processor 29 also controls the configuration of the unit inresponse to user direction. This configuration includes aspects of theCPEMTA's 24 operation, for example, enabling echo cancellation, settingVoIP packet length, configuring a codec in the DSP, etc. as known in theart. For example, the user direction may be from interaction with acommand line interface, or from SNMP data delivered through the networkfrom a remote operational support system. Other methods of systemconfiguration are known in the art and are equally applicable.

DSP 30 is used to facilitate voice data processing, such as, forexample, echo cancellation, voice compression (to conserve cablebandwidth), voice-activity detection, jitter removal, clocksynchronization and voice packetization. Multi-media signals may beprocessed according to call control instruction information, preferablynetwork-based call signaling (“NCS”), which is known in the art, thatcan take advantage of robbed-bit signaling from a DS1 link, or a PRI orCAS data link, when PRI or CAS are used instead of robbed bit signaling.After processing, call signals are formatted according to predeterminedVoIP packaging rules, preferably Ethernet. Ethernet, robbed-bitsignaling and data link signaling are known in the art.

As an example, CPEMTA device 24 may translate robbed-bit signaling intoPacketCable compliant NCS signaling messages, thereby making endpointsat PBX 2 and CO 18 appear as conventional POTS lines to call managementserver 32. Thus voice trunk channels emanating from PBX 2, for example,can be translated into PacketCable data for control by CMS 32. Inaddition, additional features can be controlled by private ManagementInformation Base variables (“MIB”), as known in the art.

By allocating IP bandwidth as required by call processing status, IPnetwork bandwidth can be used more efficiently over networks 12 and 14.For example, if only three calls are active on a DS1 trunk, only threecalls are routed by the CPEMTA through network 12 and 14, whereas withconventional equipment, bandwidth is always reserved for and occupied byall the 24 channels that compose the DS1.

Furthermore, even finer levels of bandwidth conservation are possible asone can now use VoIP techniques that have been developed for bandwidthconservation such as Voice Activity detection. For example, when a partyto a call served by PBX 2 is silent, there is no useful informationbeing sent to a caller being served by, for example, CO switch 18.However, instead of allocating upstream (with respect to the PBX)bandwidth to the channel carrying the call in an amount that wouldotherwise be needed if the caller were not silent, the bandwidth usedcan be reduced. Since there is no voice information that needs to besent from PBX 2 to the other caller, DSP 30 can recognize this. Based onthis recognition, DSP 30 can avoid sending transmitting packets thatrepresent the channel, since the packets are essentially null anyway.

At the beginning of the silence, the processor 29 in response to atrigger from the DSP 30 can generate an Ethernet packet that includes amessage conveying that until further notice, silence can be played atthe other caller's location. In practice, instead of being completelysilent, the silence message could be detected at the end receiving thesilence and instruct a noise generator to produce some white noise toimitate noise that would normally be unconsciously perceived acaller/listener. Thus, a caller would not perceive a drastic differencebetween a normal POTS call, which would include inherent backgroundnoise when the other caller (or another caller in a multi-callerconference, for example) is silent, and a VoIP call, during a time whenpackets are not transmitted due to the other's silence.

In addition to advantageously reducing bandwidth usage in various ways,device 24 supports transcoding between the continuous PCM u-law or A-lawbit streams and the various packetized VoIP codecs. Thus, externaland/or remotely located processors are not needed to perform thesefunctions. Accordingly, as discussed above, device 24 facilitates morethan mere transport of call signals from PBX 2 to other endpoints.

Turning now to FIG. 3, a flow diagram illustrating a method 300 fordelivering voice, or other multi-media content, using VoIP protocolsover networks 12 and 14 is shown. After the process begins at step 302,DS1 links are terminated at device 24, as shown in FIG. 1. This is aphysical step that involves personnel connecting a DS1 link to anappropriate connector for receiving the TDM channels from PBX 2.

At step 306, a DS1's payload is processed according to the call controlstates of the channels and any applicable VoIP provisioning, asdiscussed above. The signaling from the DS1 and the call control stateof the VoIP connections determine what happens to each channel of theDS1 payload. The processor directs the DSP to process the trafficsignals (typically PCM bytes) from a channel with an active call.However, an idle channel is dropped. The DSP performspacketizing-related voice processing and the DSP also acts on thepackets coming from the VoIP network and processes them for transmissionalong the DS1 link to the local end user. These actions may includeadjusting volume levels, or enabling packet-loss concealment, forexample, in addition to the attributes discussed previously.

At step 308, after call control processing has been performed, thepayload is packaged according to a predetermined criteria, or VoIPprotocol, such as, PacketCable, SIP or H.323. For instance, PacketCableallows VoIP packets only in 5 ms, 10 ms and 20 ms packetizations. Inaddition, the data is preferably packaged into Ethernet packets fordelivery across an IP network. At step 310, the Ethernet packets aredelivered according to the chosen VoIP addressing protocol known in theart. This facilitates applying QoS limits on certain calls, therebyproviding another way to manage bandwidth usage as compared to justproviding mere transport of calls from endpoint to endpoint. It will beappreciated that the figure illustrates steps taken in the upstreamdirection, i.e. a caller's speech signals are transmitted from the userslocation toward the network. However, similar steps may be taken at thereceiving endpoint but in a different order than those given in thefigure. In addition, a particular application may perform the steps in adifferent order even for upstream speech signals.

Turning now to FIG. 4, a flow diagram illustrates the steps in a typicalscenario 400 where the aspect is used in a PBX environment with aPacketCable CMS. It will be appreciated that the flow diagram may besimilar where VoIP protocols other that PacketCable are used, withdifferences occurring where differences in the way PacketCable and otherVoIP protocols are depicted. At step 402, the process starts and when auser picks up a telephone to place a call, the PBX indicates lineseizure (the line is no longer open) with signaling bits at step 404.The framer, which terminates the DS1 line, provides signaling bits tothe cable modem processor. These signaling bits cause the processor torecognize that an off-hook condition exists with the telephone set atstep 408, and the processor sends an off hook message to the VoIPCMS—call agent 32 as shown in FIG. 2—at step 410.

When the CMS has received and logged the off-hook message, it sends amessage to the processor at step 412, instructing it to provide a dialtone to the PBX user. The cable modem processor then instructs DSP 30,as shown in FIG. 2, to generate a dial tone and send it to the PBX atstep 414. The PBX user hears the dial tone in the telephone receiver416, and is thus informed that a call can be placed.

These and many other objects and advantages will be readily apparent toone skilled in the art from the foregoing specification when read inconjunction with the appended drawings. It is to be understood that theembodiments herein illustrated are examples only, and that the scope ofthe invention is to be defined solely by the claims when accorded a fullrange of equivalents.

1. A method for converting circuit-switched voice traffic payload datafrom a multi-channel data link into packet-switched VoIP traffic,comprising: terminating the multi-channel link at a media terminaladaptor; processing payload data at the media terminal adaptor accordingto call control instruction information; packaging the payload dataaccording to predetermined packaging rules; and transmitting the payloaddata according to a VoIP protocol.
 2. The method of claim 1 wherein thepredetermined packaging rules include instructions that any of thechannels in the multi-channel data link that are deemed not-active arenot to be packaged for transmission.
 3. The method of claim 2 wherein achannel that is not seized is deemed not active.
 4. The method of claim2 wherein a channel that is seized but is not carrying usefulinformation because a caller is silent is deemed not-active.
 5. Themethod of claim 1 wherein the call control instruction informationincludes instruction on how to compress or decompress outgoing orincoming traffic respectively.
 6. The method of claim 1 wherein the stepof packaging according to predetermined packaging rules includeconverting call traffic payload data into Ethernet packets.
 7. Themethod of claim 1 wherein the VoIP protocol is PacketCable.
 8. Themethod of claim 1 wherein the call control instruction information iscontained in robbed-bit signaling bits.
 9. The method of claim 1 whereinthe voice traffic payload data is processed according to SIP callcontrol instructions.
 10. The method of claim 1 wherein the voicetraffic payload data is processed according to H323 call controlinstructions.
 11. A system for terminating a multi-channel link andconverting information therefrom into packet data at a user device forcommunication over a network, comprising: means for interfacing with thenetwork means for terminating the multi-channel link, the terminatingmeans being coupled with the interfacing means; a first processor meansfor performing call control operations and for performing configurationoperations on the device, the first processor being operatively coupledwith the interfacing means and the terminating means; and a secondprocessor means coupled to the processor for performing voice dataprocessing, wherein the second processor is operatively coupled with thefirst processor.
 12. The system of claim 11 wherein the means forinterfacing with the network is a cable modem.
 13. The system of claim11 wherein the means for terminating the multi-channel link is a framer.14. The system of claim 13 wherein the framer is an integrated accessdevice.
 15. The system of claim 11 wherein the first processor is amicroprocessor.
 16. The system of claim 11 wherein the second processoris a digital signal processor.
 17. The system of claim 11 wherein thecall control operations include messaging to establish, maintain orrelease a call connection.
 18. The system of claim 11 wherein the cablemodem comprises the first processor.